ZyXEL Communications prestige p-2002 User Manual

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P-2002 Series User’s Guide
98
Chapter 12 Troubleshooting
12.7  Problems with Telephone or Telephone Port
12.8  Problems with Voice Service
Table 39   Troubleshooting Telephone
PROBLEM
CORRECTIVE ACTION
There is no dial 
tone or I can’t make 
or receive calls.
or
There is beeping 
instead of the dial 
tone.
Check the telephone connections and telephone wire.
Beeping means that there is not a SIP account registered for the phone to use.
You can check the Prestige’s IP addresses and VoIP status in the Maintenance 
Status screen. 
Make sure you have the VoIP screen properly configured. If you configured a SIP 
account to receive calls on only one of the phone ports, make sure your phone is 
connected to that port.
Make sure you have the Phone Port screen properly configured. If you 
configured a phone port to only use one of the SIP accounts for outgoing calls,  
make sure that SIP account is properly configured and active (see the VoIP and 
Maintenance Status screens). 
There is a beep 
before the dial tone.
A single beep before the dial tone indicates that there is a voice message for SIP 
account 1.
Two beeps before the dial tone indicate that there is a voice message for SIP 
account 2.
Use your voice service provider’s instructions to check your voice messages.
Table 40   Troubleshooting Voice Service
PROBLEM
CORRECTIVE ACTION
After the VoIP is 
configured and 
working, others are 
unable to call you 
or you lose your 
connection during a 
call. There is a NAT 
router between the 
Prestige and the 
SIP server. 
This could be caused by a short NAT UDP session timeout on the NAT router. 
When the SIP session’s entry in the NAT table times out, the NAT router does not 
have any record to use for forwarding VoIP traffic to the Prestige.
If possible, set the NAT router to use a longer NAT UDP session timeout.
Otherwise, try one of the following:
Shorten the registration expiration period (see the Expiration Duration field 
in the VoIP Advanced screen) in order to cause the Prestige to re-register 
with the SIP register server more frequently. Note that this will not help if the 
SIP register server enforces a long registration expiration period (since the 
Prestige will also use the period set by the SIP register server).
Use STUN. If your VoIP service provider does not have a STUN server, you 
can still enable STUN and enter the IP address and port number of the SIP 
server in the STUN server fields. This causes the Prestige to send STUN 
requests to the SIP server. While this will not make STUN work (since there 
won’t be any responses to the STUN requests), it should keep the NAT UDP 
session in the NAT router.